'm' - The conference is in so called monitor mode ( Only listen, no talking)'t' - The conference is in talk mode ( Talk only, no listening)'T' - Turns on the talker detection, it sends information to the manager interface and meetme list.'i' - Announce who is joining/leaving the conference.'p' - User can exit conference by pressing the # key.'X' - User can exit conference by entering a single valid extensions ( ${MEETME_EXIT_CONTEXT} ), if this variable is not set - uses extension in current context.'d' - Dynamically adds conference (Can not set pin for the conference).'D' - Dynamically adds conference (Can set pin for the conference).'e' - Selects first empty conference'E' - Selects first empty pinless conference'v' - The conference is in video mode.'r' - If this option is selected, the conference conversation will be recorded in format ${MEETME_RECORDINGFORMAT} and saved as ${MEETME_RECORDINGFILE}.'q' - Disable the enter/leave sounds.'C' - On enter announce how many users are in the conference.'M' - When there is only one user in the conference, music on hold will be played.'x' - Close conference when last marked user exits'w' - Wait until the marked user to enters the conference'b' - Run AGI script set in the ${MEETME_AGI_BACKGROUND} variable, the default script is : conf-background.agiNote: This works only with Zap channels in the same conference)

's' - Enter menu (user and administrator) when *-key is pressed'a' - Set admin mode.'A' - Set marked mode.'P' - Always ask for pin even if it is specified.

List of all used variables

${MEETME_RECORDINGFILE} - What will be the filename for the recorded conference ( option 'r' ).${MEETME_RECORDINGFORMAT} - Recording format (gsm, wav, ...).${MEETME_EXIT_CONTEXT} - The context for exit out from conference.${MEETME_AGI_BACKGROUND} - AGI script for the conference ( ZAP only ).${MEETMESECS} - Number of seconds a user participated in conference

Purpose and usage

This application is used primary for creating so called conference 'rooms'.

To see how the application works we recommend to use our IAX softphone Idefisk. You can download it from here. Please also read our tutorial to learn how to configure it to work with Asterisk PBX.

Asterisk PBX configurations

NOTE: This is only an example of what for you can use this application. Of course you can use it and for other things.

iax.conf Configurations

We need two registered users in iax.conf file. This is because we are going to use the IAX2 protocol. If you want to use other protocol such as SIP or MGCP, you have to do the configurations below respectively in sip.conf or mgcp.conf.

So, we have registered the users:userA and userB.

Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=training - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.

Now lets add and one sip user.

sip.conf Configurations

We need one registered users in sip.conf file. This is because we want to test and the SIP protocol.

So, we have registered the user sip_user.

Type=friend means that this user can make and receive calls. Host=dynamic means that the IP is not static but dynamic through a DHCP server. Allow=all means that the line which this user will use, could support all audio codecs. Context=training - this shows that this user is working with the extensions in this context of the configuration file extensions.conf.

meetme.conf Configurations

In order create conference room we have to configure it in the meetme.conf (look the picture below):

For this tutorial we are going to use meetme room number 1234, it will have also and PIN code for access (1111), and PIN code for the conference administrator (2222). We just follow the standard template:

Manish (ez8ex7yypuf at outlook dot com)25 September 2015 12:38:26Thank you for including YouTube in your list, most peploe seem to have forgotten or never known that this too is a social media slash disocvery site. People are posting comments and sending messages to each other left and right while still claiming YouTube is strictly a video platform well, of course it is, but it is also a social networkiung site, and a very useful one.

Juancho (bvninc at hotmail dot com)04 November 2008 21:03:17I know how to create Meetme rooms but using DID's but how can I enter the same conference room butfrom an IP Phone?
in other words how can I be in a confernce from an IP Phone? Wht do I ned to create or configure?

Farhad (farhad dot i at caspel dot com)22 February 2008 10:26:53Thank's good example

reloded (reloded at gmail dot com)02 March 2007 06:59:49Hi guys. Am new to asterisk. I've followed the guides here step by step and all's good so far.
However, trying to dial from x-lite, I get "Call failed: Not Found" No error code.
On the CLI on asterisk, I get
[Mar 2 08:52:20] WARNING[11507]: app_dial.c:1062 dial_exec_full: Dial argument takes format (technology/[device:]number1)

Someone please advice. I've been searching for a solution for the past 3 days with no success.

Thanks

Jo (mop77008 at mail dot telepac dot pt)27 October 2006 18:42:10I've tryed all the methods and I keep having the same warnings and error...

Andrew (asterisk at siliconandsynapse dot com)06 August 2006 21:41:37raju
You need to get the zaptel package from asterisk. Uncomment the line that says #ztdummy in the Makefile, compile and install. That will provide you with ztdummy module. Recompile asterisk. It should now detect that you have installed the zaptel package and it will create the meetme app. Then what I did, was simply copy app_meetme to my lib directory. (Its probably better if you reinstall asterisk.) Then you need to load the kernel modules zaptel and ztdummy. ztdummy provides a timer for the meetme app. Then follow the steps above to setup your extensions.conf file and meetme.conf and your good to go.

The link below is what I used to get it working
http://www.automated.it/guidetoasterisk.htm

raju (pkannamraju at hyd dot hellosoft dot com)02 June 2006 13:05:09hi,
I have set up everything as mentioned.
When i call from sip useragent(xlite) to
the conference extension i am getting 603
user declined
The error i am getting in cli prompt is: